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This document defines a mechanism by which a SIP server acting as a traditional Private Branch Exchange (PBX) can register with a SIP Service Provider (SSP) to receive phone calls for extensions designated by phone numbers. In order to function properly, this mechanism relies on the fact that the phone numbers are fully qualified and globally unique.
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1.
Introduction
2.
Constraints
3.
Terminology
4.
Mechanism Overview
5.
Registering for Multiple Phone Numbers
6.
SSP Processing of Inbound Phone Number Requests
7.
Interaction with Other Mechanisms
7.1.
Globally Routable User-Agent URIs (GRUU)
7.1.1.
Public GRUUs
7.1.2.
Temporary GRUUs
7.2.
Registration Event Package
7.2.1.
PBX Aggregate Registration State
7.2.2.
Individual Extension Registration State
7.3.
Client-Initiated (Outbound) Connections
7.4.
Non-Adjacent Contact Registration (Path)
7.5.
Service Route Discovery
8.
Examples
8.1.
Usage Scenario: Basic Registration
8.2.
Usage Scenario: Using Path to Control Request URI
9.
Requirements Analysis
10.
IANA Considerations
10.1.
New SIP Option Tag
10.2.
New SIP URI Parameters
10.2.1.
'bnc' SIP URI paramter
10.2.2.
'sg' SIP URI paramter
11.
Security Considerations
12.
References
12.1.
Normative References
12.2.
Informative References
§
Author's Address
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One of SIP's primary functions is providing rendezvous between users. By design, this rendezvous has been provided through a combination of the server look-up procedures defined in RFC 3263 [3] (Rosenberg, J. and H. Schulzrinne, “Session Initiation Protocol (SIP): Locating SIP Servers,” June 2002.), and the registrar procedures described in RFC 3261 [2] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.).
The intention of the original protocol design was that any user's AOR would be handled by the authority indicated by the hostport portion of the AOR. The users registered individual reachability information with this authority, which would then route incoming requests accordingly.
In actual deployments, some SIP servers have been deployed in architectures that, for various reasons, have requirements to provide dynamic routing information for large blocks of AORs, where all of the AORs in the block were to be handled by the same server. For purposes of efficiency, many of these deployments do not wish to maintain separate registrations for each of the AORs in the block. This leads to the desire for an alternate mechanism for providing dynamic routing information for blocks of AORs.
Because this problem has certain similarities with the REGISTER operation, several non-standard, ad hoc extensions to REGISTER have been developed to address this desire.
Although the use of REGISTER to update reachability information for multiple users simultaneously is somewhat beyond the original semantics defined for REGISTER, this approach has seen significant deployment in certain environments. In particular, deployments in which small to medium SIP PBX servers are addressed using E.164 numbers have used this mechanism to avoid the need to maintain DNS entries or static IP addresses for the PBX servers.
In recognition of the momentum that a REGISTER-based approach has within that relatively narrow ecological niche, this document defines a REGISTER-based approach that is tailored to E.164-addressed extensions in a SIP PBX environment. It is not intended for general-purpose registration of SIP URIs in which the user portion is non-numeric or non-globally-unique.
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The following paragraph is perhaps the most important in understanding the solution defined in this document.
Within the problem space that has been established for this work, several constraints shape our solution. These are being defined in the MARTINI requirements document [5] (Elwell, J. and H. Kaplan, “Requirements for multiple address of record (AOR) reachability information in the Session Initiation Protocol (SIP),” March 2010.). In terms of impact to the solution at hand, the following two constraints have the most profound effect: (1) The PBX cannot be assumed to be assigned a static IP address; and (2) No DNS entry can be relied upon to consistently resolve to the IP address of the PBX.
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The key words "MUST", "MUST NOT", "REQUIRED", "SHALL","SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [1] (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.).
Further, the term "SSP" is meant as an acronym for a "SIP Service Provider," while the term "PBX" is used to indicate a SIP Private Branch Exchange.
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The overall mechanism is achieved using a REGISTER request with a specially-formatted Contact URI. This document also defines an option tag that can be used to ensure a registrar and any intermediaries understand the mechanism described herein.
The Contact URI itself is tagged with a URI parameter to indicate that it actually represents a multitude of phone-number-associated contacts.
We also define some lightweight extensions for GRUU to allow the use of public and temporary GRUUs assigned by the SSP.
Aside from these extensions, the REGISTER message itself is processed by a registrar in the same way as normal registrations: by updating its location service with additional AOR to Contact bindings.
Note that the list of extensions associated with a PBX is a matter of local provisioning at the SSP and at the PBX. The mechanism defined in this document does not provide any means to detect or recover from provisioning mismatches (although the registration event package can be used as a standardized means for auditing such extensions; see Section 7.2.1 (PBX Aggregate Registration State)).
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To register for multiple phone numbers, the PBX sends a REGISTER message to the SSP. This REGISTER varies from a typical register in two important ways. First, it must contain an option tag of "bulknumbercontact" in both a "Require" header field and a "Proxy-Require" header field. Second, in at least one "Contact" header field, it must include a Contact URI that contains the URI parameter "bnc", and no user portion (hence no "@" symbol). A URI with a "bnc" parameter MUST NOT contain a user portion.
Because of the constraints discussed in Section 2 (Constraints), the host portion of the Contact URI will generally contain an IP address, although nothing in this mechanism enforces or relies upon that fact. If the PBX operator chooses to maintain DNS entries that resolve to the IP address of his PBX via RFC 3263 resolution procedures, then this mechanism works just fine with domain names in the Contact header field.
The URI parameter indicates that special interpretation of the Contact URI is necessary: instead of representing a single, concrete Contact URI to be inserted into the location service, it represents a multitude of Contact URIs (one for each associated phone numbers), semantically resulting in a multitude of AOR-to-Contact rows in the location service.
The registrar, upon receipt of a REGISTER message in the foregoing form, will use the value in the "To" header field to identify the PBX for which registration is being requested. It then authenticates the PBX (using, e.g., SIP Digest authentication, mutual TLS, or some other authentication mechanism). After the PBX is authenticated, the registrar updates its location service so that each of the phone numbers associated with the PBX creates a unique AOR to Contact mapping. Semantically, each of these mappings will be treated as a unique row in the location service. The actual implementation may, of course, perform internal optimizations to reduce the amount of memory used to store such information.
For each of these unique rows, the AOR will be in the format that the SSP expects to receive from external parties (e.g. "sip:+12145550102@ssp.example.com"), and the corresponding Contact will be formed adding a user portion to the REGISTER's Contact URI containing the fully-qualified, E.164-formatted phone number (including the preceding "+" symbol) and removing the "bnc" parameter. For example, if the "Contact" header field contains the URI <sip:198.51.100.3:5060;user=phone;bnc>, then the Contact value associated with the aforementioned AOR will be <sip:+12145550102@198.51.100.3:5060;user=phone>.
Aside from the "bnc" parameter, all URI parameters present on the "Contact" URI in the REGISTER message MUST be copied to the Contact value stored in the location service.
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In general, after processing the AOR to Contact mapping described in the preceding section, the SSP Proxy/Registrar (or equivalent entity) performs traditional Proxy/Registrar behavior, based on the mapping. For inbound SIP requests whose AOR indicates an E.164 number assigned to one of the SSP's customers, this will generally involve setting the target set to the registered contacts associated with that AOR, and performing request forwarding as described in section 16.6 of RFC 3261 [2] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.).
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The following sections describe the means by which this mechanism interacts with relevant REGISTER-related extensions currently defined by the IETF.
Currently, the descriptions are somewhat informal, and omit some details for the sake of brevity. If the MARTINI working group expresses interest in furthering the mechanism described by this document, they will be fleshed out with more detail and formality.
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To enable advanced services to work with extensions behind a SIP PBX, it is important that the GRUU mechanism defined by RFC 5627 [10] (Rosenberg, J., “Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP),” October 2009.) work correctly with the mechanism defined by this document.
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When a PBX registers a Bulk Number Contact (a Contact with a "bnc" parameter), and also invokes GRUU procedures for that Contact during registration, then the SSP will assign a public GRUU to the PBX in the normal fashion. Because the URI being registered contains a "bnc" parameter, the GRUU will also contain a "bnc" parameter. In particular, this means that the GRUU will not contain a user portion.
When a terminal registers with the PBX using GRUU procedures for a Contact, it adds an "sg" parameter to the GRUU parameter it received from the SSP. This "sg" parameter contains a disambiguation token that the SSP can use to route the request to the proper user agent.
So, for example, when the PBX registers with the following contact header field:
Contact: <sip:198.51.100.3;user=phone;bnc>; +sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
Then the SSP may choose to respond with a Contact header field that looks like this:
<allOneLine> Contact: <sip:198.51.100.3;user=phone;bnc>; pub-gruu="sip:ssp.example.com;gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6"; +sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>" ;expires=7200 </allOneLine>
When its own terminals register, the PBX can then add whatever device identifier it feels appropriate in an "sg" parameter, and present this value to its own terminals. For example, assume the extension associated with the phone number "+12145550102" sent the following Contact header field in its register:
Contact: <sip:line-1@10.20.1.17>; +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
The PBX will add an "sg" parameter to the pub-gruu it received from the SSP with a token that uniquely identifies the device (possibly the URN itself; possibly some other identifier); insert a user portion containing the fully-qualified E.164 number associated with the extension; and return the result to the terminal as its public GRUU. The resulting Contact header field would look something like this:
<allOneLine> Contact: <sip:line-1@10.20.1.17>; pub-gruu="sip:+12145550102@ssp.example.com;gr=urn: uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6;sg=00:05:03:5e:70:a6"; +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>" ;expires=3600 </allOneLine>
When an incoming request arrives at the SSP for a GRUU corresponding to a bulk number contact ("bnc"), the SSP performs slightly different processing for the GRUU than a Proxy/Registrar would. When the GRUU is re-targeted to the registered bulk number contact, the SSP MUST copy the "sg" parameter from the GRUU to the new target. The PBX can then use this "sg" parameter to determine which user agent the request should be routed to.
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PBXes have two options for creating temporary GRUUs for use by its terminals.
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If a PBX wishes to provide temporary GRUUs for its terminals, it may do so by producing its own "Self-made GRUUs" (as defined in section 4.3 of RFC 5627 [10] (Rosenberg, J., “Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP),” October 2009.)). These GRUUs are produced using the PBX's own IP address (or domain, if it maintains one in DNS). The temporary GRUUs are then propagated to terminals using normal GRUU mechanism.
The ability to produce temporary GRUUs in this fashion is predicated on the conditions described in section 4.3 of RFC 5627. In particular, it requires PBX to be publicly routable, and willing to accept requests destined for its own Self-made GRUUs from sources other than the SSP. If these conditions cannot be satisfied (or the PBX operator chooses not to satisfy them for policy reasons), then the PBX users will not be able to make use of temporary GRUUs.
This mechanism is also predicated on the IP address for the PBX being relatively stable over a long period of time. This is generally a safe assumption to make, as frequent PBX IP address changes will result in intermittent connectivity issues and interruptions to ongoing calls.
On a related note: when used with this extension, the SSP will not return a temporary GRUU in the registration response for any contacts that include a "bnc" parameter in their URI.
For example, using the same setup as in the "Public GRUU" section above, an extensions registering with the PBX might obtain a temp gruu by receiving a Contact header field that looks like:
<allOneLine> Contact: <sip:line-1@10.20.1.17>; pub-gruu="sip:ssp.example.com;gr=urn:uuid:f81d4fae-7dec-11d0-a765- 00a0c91e6bf6;sg=a0471c99573b877b"; +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>" ;expires=3600 </allOneLine>
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If a PBX does not satisfy the criteria for producing its own "Self-made GRUUs," then it may create temporary GRUUs based on the public GRUUs it received from the SSP at registration time. To create Temporary GRUUs of this form, the PBX will add an opaque "sg" parameter to the public GRUU it received from the SSP, and will omit the user portion.
Note that, because these GRUUs are temporary GRUUs, a unique "sg" parameter will be generated for each successful registration attempt. The PBX tracks the various "sg" values associated with each user agent, and can re-target to the correct instance when the request arrives.
For this approach to function, the SSP must be able to resolve a GRUU based solely on the value of its "gr" parameter, as the user portion of the GRUU will not contain an E.164 number. Further, the SSP will not know which actual extension the request is destined for, only that it corresponds to an extension belonging to the PBX.
Using the same basic setup as the example for the public GRUU, a terminal might receive a temporary GRUU by getting back a Contact header field that looks like this:
<allOneLine> Contact: <sip:line-1@10.20.1.17>; temp-gruu="sip:ssp.example.com;gr=urn:uuid:f81d4fae-7dec-11d0-a765- 00a0c91e6bf6;sg=0UYYRV046P";+sip.instance="<urn:uuid:d0e2f290-104b- 11df-8a39-0800200c9a66>";expires=3600 </allOneLine>
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As this mechanism inherently deals with REGISTER behavior, it is imperative to consider its impact on the Registration Event Package defined by RFC 3680 [8] (Rosenberg, J., “A Session Initiation Protocol (SIP) Event Package for Registrations,” March 2004.). In practice, there will be two main use cases for subscribing to registration data: learning about the overall registration state for the PBX, and learning about the registration state for a single PBX extension.
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If the PBX (or another interested and authorized party) wishes to monitor or audit the registration state for all of the extensions currently registered to that PBX, it can subscribe to the SIP registration event package at the PBX's main URI -- that is, the URI used in the "To" header field of the REGISTER message.
The NOTIFY messages for such a subscription will contain a body that contains one record for each phone number associated with the PBX. The AORs will be in the format expected to be received by the SSP (e.g., "sip:+12145550105@ssp.example.com"), and the Contacts will correspond to the mapped Contact created by the registration (e.g., "sip:+12145550105@98.51.100.3").
In particular, the "bnc" parameter is forbidden from appearing in the body of a reg-event notify.
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If the SSP receives a SUBSCRIBE request for the registration event package with a Request-URI that indicates a contact registered via the "Bulk Number Contact" mechanism defined in this document, then it MUST proxy that SUBSCRIBE to the PBX in the same way that is would proxy an INVITE bound for that AOR.
Defining the behavior in this way is important, since the reg-event subscriber is interested in finding out about the comprehensive list of devices associated with the phone number. Only the PBX will have authoritative access to this information. For example, if the user has registered multiple terminals with differing capabilities, the SSP will not know about the devices or their capabilities. By contrast, the PBX will.
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RFC 5626 [9] (Jennings, C., Mahy, R., and F. Audet, “Managing Client-Initiated Connections in the Session Initiation Protocol (SIP),” October 2009.) -- needs analysis. Some people think it might "just work."
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RFC 3327 [6] (Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” December 2002.) -- needs analysis. Some people think it might "just work."
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RFC 3608 [7] (Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration,” October 2003.) -- needs analysis. Some people think it might "just work."
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These will be fleshed out more in later versions of the draft, with explanations of the processing performed at each step. For the time being, they just show the basic syntax described above.
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This example shows a basic bulk REGISTER transaction, followed by an INVITE addressed to one of the registered terminals.
Internet SSP PBX | | | | |REGISTER | | |Contact:<sip:198.51.100.3;bnc> | | |<--------------------------------| | | | | |200 OK | | |-------------------------------->| | | | |INVITE | | |sip:+12145550105@ssp.example.com| | |------------------------------->| | | | | | |INVITE | | |sip:+12145550105@198.51.100.3 | | |-------------------------------->|
REGISTER sip:ssp.example.com SIP/2.0 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: <sip:pbx@ssp.example.com> From: <sip:pbx@ssp.example.com>;tag=a23589 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Require: bulknumbercontact Contact: <sip:198.51.100.3:5060;user=phone;bnc> Expires: 7200 Content-Length: 0
INVITE sip:+12145550105@ssp.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad Max-Forwards: 69 To: <sip:2145550105@some-other-place.example.net> From: <sip:gsmith@example.org>;tag=456248 Call-ID: f7aecbfc374d557baf72d6352e1fbcd4 CSeq: 24762 INVITE Contact: <sip:line-1@192.0.2.178:2081> Content-Type: application/sdp Content-Length: ... <sdp body here>
INVITE sip:+12145550105@198.51.100.3;user=phone SIP/2.0 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50 Max-Forwards: 68 To: <sip:2145550105@some-other-place.example.net> From: <sip:gsmith@example.org>;tag=456248 Call-ID: 7ca24b9679ffe9aff87036a105e30d9b CSeq: 24762 INVITE Contact: <sip:line-1@192.0.2.178:2081> Content-Type: application/sdp Content-Length: ... <sdp body here>
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This example shows a bulk REGISTER transaction with the SSP making use of the "Path" header field extension [6] (Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” December 2002.). This allows the SSP to designate a domain on the incoming Request URI that does not necessarily resolve to the PBX from when the SSP applies RFC 3263 procedures to it.
Internet SSP PBX | | | | |REGISTER | | |Path:<sip:pbx@198.51.100.3;lr> | | |Contact:<sip:pbx.example;bnc> | | |<--------------------------------| | | | | |200 OK | | |-------------------------------->| | | | |INVITE | | |sip:+12145550105@ssp.example.com| | |------------------------------->| | | | | | |INVITE | | |sip:+12145550105@pbx.example | | |Route:<sip:pbx@198.51.100.3;lr> | | |-------------------------------->|
REGISTER sip:ssp.example.com SIP/2.0 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: <sip:pbx@ssp.example.com> From: <sip:pbx@ssp.example.com>;tag=a23589 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Require: bulknumbercontact Path: <sip:pbx@198.51.100.3:5060;lr> Contact: <sip:pbx.example;user=phone;bnc> Expires: 7200 Content-Length: 0
INVITE sip:+12145550105@ssp.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad Max-Forwards: 69 To: <sip:2145550105@some-other-place.example.net> From: <sip:gsmith@example.org>;tag=456248 Call-ID: f7aecbfc374d557baf72d6352e1fbcd4 CSeq: 24762 INVITE Contact: <sip:line-1@192.0.2.178:2081> Content-Type: application/sdp Content-Length: ... <sdp body here>
INVITE sip:+12145550105@pbx.example;user=phone SIP/2.0 Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50 Route: <sip:pbx@198.51.100.3:5060;lr> Max-Forwards: 68 To: <sip:2145550105@some-other-place.example.net> From: <sip:gsmith@example.org>;tag=456248 Call-ID: 7ca24b9679ffe9aff87036a105e30d9b CSeq: 24762 INVITE Contact: <sip:line-1@192.0.2.178:2081> Content-Type: application/sdp Content-Length: ... <sdp body here>
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The document "Requirements for multiple address of record (AOR) reachability information in the Session Initiation Protocol (SIP)" [5] (Elwell, J. and H. Kaplan, “Requirements for multiple address of record (AOR) reachability information in the Session Initiation Protocol (SIP),” March 2010.) contains a list of requirements and desired properties for a mechanism to register multiple AORs with a single SIP transaction. This section evaluates those requirements against the mechanism described in this document.
REQ1 - The mechanism MUST allow a SIP-PBX to enter into a trunking arrangement with an SSP whereby the two parties have agreed on a set of telephone numbers deemed to have been assigned to the SIP-PBX.
The requirement is satisfied.
REQ2 - The mechanism MUST allow a set of assigned telephone numbers to comprise E.164 numbers, which can be in contiguous ranges, discrete, or in any combination of the two.
The requirement is satisfied; the DIDs associated with a registration is established by bilateral agreement between the SSP and the PBX, and is not part of the mechanism described in this document.
REQ3 - The mechanism MUST allow a SIP-PBX to register reachability information with its SSP, in order to enable the SSP to route to the SIP-PBX inbound requests targeted at assigned telephone numbers.
The requirement is satisfied.
REQ4 - The mechanism MUST NOT prevent UAs attached to a SIP-PBX registering with the SIP-PBX on behalf of AORs based on assigned telephone numbers in order to receive requests targeted at those telephone numbers, without needing to involve the SSP in the registration process.
The requirement is satisfied; in the presumed architecture, PBX terminals register with the PBX, an require no interaction with the SSP.
REQ5 - The mechanism MUST allow a SIP-PBX to handle internally requests originating at its own UAs and targeted at its assigned telephone numbers, without routing those requests to the SSP.
The requirement is satisfied; PBXes may recognize their own DID and their own GRUUs, and perform on-PBX routing without sending the requests to the SSP.
REQ6 - The mechanism MUST allow a SIP-PBX to receive requests to its assigned telephone numbers originating outside the SIP-PBX and arriving via the SSP, so that the PBX can route those requests onwards to its UAs, as it would for internal requests to those telephone numbers.
The requirement is satisfied
REQ7 - The mechanism MUST provide a means whereby a SIP-PBX knows which of its assigned telephone numbers an inbound request from its SSP is targeted at.
The requirement is satisfied. For ordinary calls and calls using Public GRUUs, the DID is indicated in the user portion of the Request-URI. For calls using Temp GRUUs constructed with the mechanism described in Section 7.1.2.2 (Approach 2 - Anonymous Public GRUUs), the "sg" parameter provides a correlation token the PBX can use to identify which terminal the call should be routed to.
REQ8 - The mechanism MUST provide a means of avoiding problems due to one side using the mechanism and the other side not.
The requirement is satisfied through the 'bulknumbercontact' option tag and the 'bnc' Contact parameter.
REQ9 - The mechanism MUST observe SIP backwards compatibility principles.
The requirement is satisfied through the 'bulknumbercontact' option tag.
REQ10 - The mechanism MUST work in the presence of intermediate SIP entities on the SSP side of the SIP-PBX-to-SSP interface (i.e., between the SIP-PBX and the SSP's domain proxy), where those intermediate SIP entities need to be on the path of inbound requests to the PBX.
The requirement is satisfied through the use of the Path mechanism defined in RFC 3327 [6] (Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” December 2002.)
REQ11 - The mechanism MUST work when a SIP-PBX obtains its IP address dynamically.
The requirement is satisfied by allowing the PBX to use an IP address in the Bulk Number Contact URI contained in a REGISTER Contact header field.
REQ12 - The mechanism MUST work without requiring the SIP-PBX to have a domain name or the ability to publish its domain name in the DNS.
The requirement is satisfied by allowing the PBX to use an IP address in the Bulk Number Contact URI contained in a REGISTER Contact header field.
REQ13 - For a given SIP-PBX and its SSP, there MUST be no impact on other domains, which are expected to be able to use normal RFC 3263 procedures to route requests, including requests needing to be routed via the SSP in order to reach the SIP-PBX.
The requirement is satisfied by allowing the domain name in the Request URI used by external entities to resolve to the SSP's servers via normal RFC 3263 resolution procedures.
REQ14 - The mechanism MUST be able to operate over a transport that provides integrity protection and confidentiality.
The requirement is satisfied; nothing in the proposed mechanism prevent the use of TLS between the SSP and the PBX.
REQ15 - The mechanism MUST support authentication of the SIP-PBX by the SSP and vice versa.
The requirement is satisfied; PBXes may employ either SIP digest authentication or mutually-authenticated TLS for authentication purposes.
REQ16 - The mechanism MUST allow the SIP-PBX to provide its UAs with public or temporary Globally Routable UA URIs (GRUUs) [10] (Rosenberg, J., “Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP),” October 2009.).
The requirement is satisfied via the mechanisms detailed in Section 7.1 (Globally Routable User-Agent URIs (GRUU)).
REQ17 - The mechanism MUST NOT preclude the ability of the SIP-PBX to route on-PBX requests directly, without hair-pinning the signaling through the SSP.
The requirement is satisfied; PBXes may recognize their own DID and their own GRUUs, and perform on-PBX routing without sending the requests to the SSP. (Note that this requirement duplicates REQ5, and will probably be removed in a future version of the requirements document.)
REQ18 - The mechanism MUST work over any existing transport specified for SIP, including UDP.
The requirement is satisfied to the extent that UDP can be used for REGISTER requests in general. The application of certain extensions and/or network topologies may exceed UDP MTU sizes, but such issues arise both with and without the mechanism described in this document. This document does not exacerbate such issues.
DES1 - The mechanism SHOULD allow an SSP to exploit its mechanisms for providing SIP service to ordinary subscribers in order to provide a SIP trunking service to SIP-PBXes.
The desired property is satisfied; the routing mechanism described in this document is identical to the routing performed for singly-registered AORs.
DES2 - The mechanism SHOULD scale to SIP-PBX's of several thousand assigned telephone numbers.
The desired property is satisfied; nothing in this document precludes DID pools of arbitrary size.
DES3 - The mechanism SHOULD scale to support several thousand SIP-PBX's on a single SSP.
The desired property is satisfied; nothing in this document precludes an arbitrary number of PBXes from attaching to a single SSP.
DES4 - The mechanism SHOULD require relatively modest changes to a substantial population of existing SSP and SIP-PBX implementations, in order to encourage a fast market adoption of the standardized mechanism.
The desired property is difficult to evaluate in the context of any solution. The mechanism proposed in this document uses the REGISTER method, which is the method preferred by many existing PBX deployments. The handling of request routing logic is nearly identical to that of RFC 3261 proxy/registrars, allowing implementors to leverage existing proxy/registrar code.
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This document registers a new SIP option tag to indicate support for the mechanism it defines, plus two new SIP URI parameters.
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This section defines a new SIP option tag per the guidelines in Section 27.1 of RFC 3261[2] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.).
- Name:
- bulknumbercontact
- Description:
- This option tag is used to identify the extension that provides Registration for Multiple Phone Numbers in SIP. When present in a Require or Proxy-Require header field of a REGISTER request, it indicates that support for this extension is required of registrars and proxies, respectively, that are a party to the registration transaction.
- Reference:
- RFCXXXX (this document)
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This specification defines two new SIP URI parameters, as per the registry created by RFC 3969 [4] (Camarillo, G., “The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP),” December 2004.).
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- Parameter Name:
- bnc
- Predefined Values:
- No (no values are allowed)
- Reference:
- RFCXXXX (this document)
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- Parameter Name:
- sg
- Predefined Values:
- No
- Reference:
- RFCXXXX (this document)
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There are certainly security implications associated with the mechanisms described in this document, mostly dealing with the unprecedented semantic inclusion of multiple AORs in a single REGISTER request. This section will be formulated following an analysis of the security impact of GIN on Path, Service-Route, and Outbound.
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| [1] | Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” BCP 14, RFC 2119, March 1997 (TXT, HTML, XML). |
| [2] | Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” RFC 3261, June 2002 (TXT). |
| [3] | Rosenberg, J. and H. Schulzrinne, “Session Initiation Protocol (SIP): Locating SIP Servers,” RFC 3263, June 2002 (TXT). |
| [4] | Camarillo, G., “The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP),” BCP 99, RFC 3969, December 2004 (TXT). |
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| [5] | Elwell, J. and H. Kaplan, “Requirements for multiple address of record (AOR) reachability information in the Session Initiation Protocol (SIP),” draft-ietf-martini-reqs-02 (work in progress), March 2010 (TXT). |
| [6] | Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” RFC 3327, December 2002 (TXT). |
| [7] | Willis, D. and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration,” RFC 3608, October 2003 (TXT). |
| [8] | Rosenberg, J., “A Session Initiation Protocol (SIP) Event Package for Registrations,” RFC 3680, March 2004 (TXT). |
| [9] | Jennings, C., Mahy, R., and F. Audet, “Managing Client-Initiated Connections in the Session Initiation Protocol (SIP),” RFC 5626, October 2009 (TXT). |
| [10] | Rosenberg, J., “Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP),” RFC 5627, October 2009 (TXT). |
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| Adam Roach | |
| Tekelec | |
| 17210 Campbell Rd. | |
| Suite 250 | |
| Dallas, TX 75252 | |
| US | |
| Email: | adam@nostrum.com |